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-rw-r--r--src/OutputList.cxx4
-rw-r--r--src/mixer/OssMixerPlugin.cxx (renamed from src/mixer/oss_mixer_plugin.c)16
-rw-r--r--src/output/AlsaOutputPlugin.cxx (renamed from src/output/alsa_output_plugin.c)150
-rw-r--r--src/output/AlsaOutputPlugin.hxx (renamed from src/output/alsa_output_plugin.h)6
-rw-r--r--src/output/OssOutputPlugin.cxx (renamed from src/output/oss_output_plugin.c)64
-rw-r--r--src/output/OssOutputPlugin.hxx (renamed from src/output/oss_output_plugin.h)6
-rw-r--r--src/pcm_export.h8
7 files changed, 137 insertions, 117 deletions
diff --git a/src/OutputList.cxx b/src/OutputList.cxx
index a9a0b3d33..87e441757 100644
--- a/src/OutputList.cxx
+++ b/src/OutputList.cxx
@@ -20,7 +20,7 @@
#include "config.h"
#include "OutputList.hxx"
#include "output_api.h"
-#include "output/alsa_output_plugin.h"
+#include "output/AlsaOutputPlugin.hxx"
#include "output/ao_output_plugin.h"
#include "output/ffado_output_plugin.h"
#include "output/fifo_output_plugin.h"
@@ -29,7 +29,7 @@
#include "output/mvp_output_plugin.h"
#include "output/null_output_plugin.h"
#include "output/openal_output_plugin.h"
-#include "output/oss_output_plugin.h"
+#include "output/OssOutputPlugin.hxx"
#include "output/osx_output_plugin.h"
#include "output/pipe_output_plugin.h"
#include "output/pulse_output_plugin.h"
diff --git a/src/mixer/oss_mixer_plugin.c b/src/mixer/OssMixerPlugin.cxx
index 608f1f9b8..490a65414 100644
--- a/src/mixer/oss_mixer_plugin.c
+++ b/src/mixer/OssMixerPlugin.cxx
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -206,11 +206,11 @@ oss_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r)
}
const struct mixer_plugin oss_mixer_plugin = {
- .init = oss_mixer_init,
- .finish = oss_mixer_finish,
- .open = oss_mixer_open,
- .close = oss_mixer_close,
- .get_volume = oss_mixer_get_volume,
- .set_volume = oss_mixer_set_volume,
- .global = true,
+ oss_mixer_init,
+ oss_mixer_finish,
+ oss_mixer_open,
+ oss_mixer_close,
+ oss_mixer_get_volume,
+ oss_mixer_set_volume,
+ true,
};
diff --git a/src/output/alsa_output_plugin.c b/src/output/AlsaOutputPlugin.cxx
index d8b184273..4d9f259ad 100644
--- a/src/output/alsa_output_plugin.c
+++ b/src/output/AlsaOutputPlugin.cxx
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -18,7 +18,7 @@
*/
#include "config.h"
-#include "alsa_output_plugin.h"
+#include "AlsaOutputPlugin.hxx"
#include "output_api.h"
#include "mixer_list.h"
#include "pcm_export.h"
@@ -26,6 +26,8 @@
#include <glib.h>
#include <alsa/asoundlib.h>
+#include <string>
+
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "alsa"
@@ -43,14 +45,16 @@ enum {
typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
snd_pcm_uframes_t size);
-struct alsa_data {
+struct AlsaOutput {
struct audio_output base;
- struct pcm_export_state export;
+ struct pcm_export_state pcm_export;
- /** the configured name of the ALSA device; NULL for the
- default device */
- char *device;
+ /**
+ * The configured name of the ALSA device; empty for the
+ * default device
+ */
+ std::string device;
/** use memory mapped I/O? */
bool use_mmap;
@@ -101,6 +105,18 @@ struct alsa_data {
* The number of frames written in the current period.
*/
snd_pcm_uframes_t period_position;
+
+ AlsaOutput():mode(0), writei(snd_pcm_writei) {
+ }
+
+ bool Init(const config_param *param, GError **error_r) {
+ return ao_base_init(&base, &alsa_output_plugin,
+ param, error_r);
+ }
+
+ void Deinit() {
+ ao_base_finish(&base);
+ }
};
/**
@@ -113,24 +129,13 @@ alsa_output_quark(void)
}
static const char *
-alsa_device(const struct alsa_data *ad)
-{
- return ad->device != NULL ? ad->device : default_device;
-}
-
-static struct alsa_data *
-alsa_data_new(void)
+alsa_device(const AlsaOutput *ad)
{
- struct alsa_data *ret = g_new(struct alsa_data, 1);
-
- ret->mode = 0;
- ret->writei = snd_pcm_writei;
-
- return ret;
+ return ad->device.empty() ? default_device : ad->device.c_str();
}
static void
-alsa_configure(struct alsa_data *ad, const struct config_param *param)
+alsa_configure(AlsaOutput *ad, const struct config_param *param)
{
ad->device = config_dup_block_string(param, "device", NULL);
@@ -161,10 +166,10 @@ alsa_configure(struct alsa_data *ad, const struct config_param *param)
static struct audio_output *
alsa_init(const struct config_param *param, GError **error_r)
{
- struct alsa_data *ad = alsa_data_new();
+ AlsaOutput *ad = new AlsaOutput();
- if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
- g_free(ad);
+ if (!ad->Init(param, error_r)) {
+ delete ad;
return NULL;
}
@@ -176,12 +181,10 @@ alsa_init(const struct config_param *param, GError **error_r)
static void
alsa_finish(struct audio_output *ao)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
-
- ao_base_finish(&ad->base);
+ AlsaOutput *ad = (AlsaOutput *)ao;
- g_free(ad->device);
- g_free(ad);
+ ad->Deinit();
+ delete ad;
/* free libasound's config cache */
snd_config_update_free_global();
@@ -190,18 +193,18 @@ alsa_finish(struct audio_output *ao)
static bool
alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
+ AlsaOutput *ad = (AlsaOutput *)ao;
- pcm_export_init(&ad->export);
+ pcm_export_init(&ad->pcm_export);
return true;
}
static void
alsa_output_disable(struct audio_output *ao)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
+ AlsaOutput *ad = (AlsaOutput *)ao;
- pcm_export_deinit(&ad->export);
+ pcm_export_deinit(&ad->pcm_export);
}
static bool
@@ -349,7 +352,8 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
{
/* try the input format first */
- int err = alsa_output_try_format(pcm, hwparams, audio_format->format,
+ int err = alsa_output_try_format(pcm, hwparams,
+ sample_format(audio_format->format),
packed_r, reverse_endian_r);
/* if unsupported by the hardware, try other formats */
@@ -383,15 +387,11 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
* the configured settings and the audio format.
*/
static bool
-alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
+alsa_setup(AlsaOutput *ad, struct audio_format *audio_format,
bool *packed_r, bool *reverse_endian_r, GError **error)
{
- snd_pcm_hw_params_t *hwparams;
- snd_pcm_sw_params_t *swparams;
unsigned int sample_rate = audio_format->sample_rate;
unsigned int channels = audio_format->channels;
- snd_pcm_uframes_t alsa_buffer_size;
- snd_pcm_uframes_t alsa_period_size;
int err;
const char *cmd = NULL;
int retry = MPD_ALSA_RETRY_NR;
@@ -401,6 +401,7 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
period_time_ro = period_time = ad->period_time;
configure_hw:
/* configure HW params */
+ snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
cmd = "snd_pcm_hw_params_any";
err = snd_pcm_hw_params_any(ad->pcm, hwparams);
@@ -434,7 +435,7 @@ configure_hw:
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support format %s: %s",
alsa_device(ad),
- sample_format_to_string(audio_format->format),
+ sample_format_to_string(sample_format(audio_format->format)),
snd_strerror(-err));
return false;
}
@@ -525,11 +526,13 @@ configure_hw:
if (retry != MPD_ALSA_RETRY_NR)
g_debug("ALSA period_time set to %d\n", period_time);
+ snd_pcm_uframes_t alsa_buffer_size;
cmd = "snd_pcm_hw_params_get_buffer_size";
err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
if (err < 0)
goto error;
+ snd_pcm_uframes_t alsa_period_size;
cmd = "snd_pcm_hw_params_get_period_size";
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
NULL);
@@ -537,6 +540,7 @@ configure_hw:
goto error;
/* configure SW params */
+ snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
cmd = "snd_pcm_sw_params_current";
@@ -586,7 +590,7 @@ error:
}
static bool
-alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
+alsa_setup_dsd(AlsaOutput *ad, struct audio_format *audio_format,
bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
GError **error_r)
{
@@ -626,7 +630,7 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
}
static bool
-alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
+alsa_setup_or_dsd(AlsaOutput *ad, struct audio_format *audio_format,
GError **error_r)
{
bool shift8 = false, packed, reverse_endian;
@@ -642,8 +646,9 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
if (!success)
return false;
- pcm_export_open(&ad->export,
- audio_format->format, audio_format->channels,
+ pcm_export_open(&ad->pcm_export,
+ sample_format(audio_format->format),
+ audio_format->channels,
dsd_usb, shift8, packed, reverse_endian);
return true;
}
@@ -651,12 +656,10 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
static bool
alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
- int err;
- bool success;
+ AlsaOutput *ad = (AlsaOutput *)ao;
- err = snd_pcm_open(&ad->pcm, alsa_device(ad),
- SND_PCM_STREAM_PLAYBACK, ad->mode);
+ int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
+ SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"Failed to open ALSA device \"%s\": %s",
@@ -667,20 +670,20 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
g_debug("opened %s type=%s", snd_pcm_name(ad->pcm),
snd_pcm_type_name(snd_pcm_type(ad->pcm)));
- success = alsa_setup_or_dsd(ad, audio_format, error);
- if (!success) {
+ if (!alsa_setup_or_dsd(ad, audio_format, error)) {
snd_pcm_close(ad->pcm);
return false;
}
ad->in_frame_size = audio_format_frame_size(audio_format);
- ad->out_frame_size = pcm_export_frame_size(&ad->export, audio_format);
+ ad->out_frame_size = pcm_export_frame_size(&ad->pcm_export,
+ audio_format);
return true;
}
static int
-alsa_recover(struct alsa_data *ad, int err)
+alsa_recover(AlsaOutput *ad, int err)
{
if (err == -EPIPE) {
g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
@@ -719,7 +722,7 @@ alsa_recover(struct alsa_data *ad, int err)
static void
alsa_drain(struct audio_output *ao)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
+ AlsaOutput *ad = (AlsaOutput *)ao;
if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
return;
@@ -753,7 +756,7 @@ alsa_drain(struct audio_output *ao)
static void
alsa_cancel(struct audio_output *ao)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
+ AlsaOutput *ad = (AlsaOutput *)ao;
ad->period_position = 0;
@@ -763,7 +766,7 @@ alsa_cancel(struct audio_output *ao)
static void
alsa_close(struct audio_output *ao)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
+ AlsaOutput *ad = (AlsaOutput *)ao;
snd_pcm_close(ad->pcm);
}
@@ -772,11 +775,11 @@ static size_t
alsa_play(struct audio_output *ao, const void *chunk, size_t size,
GError **error)
{
- struct alsa_data *ad = (struct alsa_data *)ao;
+ AlsaOutput *ad = (AlsaOutput *)ao;
assert(size % ad->in_frame_size == 0);
- chunk = pcm_export(&ad->export, chunk, size, &size);
+ chunk = pcm_export(&ad->pcm_export, chunk, size, &size);
assert(size % ad->out_frame_size == 0);
@@ -789,7 +792,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
% ad->period_frames;
size_t bytes_written = ret * ad->out_frame_size;
- return pcm_export_source_size(&ad->export,
+ return pcm_export_source_size(&ad->pcm_export,
bytes_written);
}
@@ -803,17 +806,20 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
}
const struct audio_output_plugin alsa_output_plugin = {
- .name = "alsa",
- .test_default_device = alsa_test_default_device,
- .init = alsa_init,
- .finish = alsa_finish,
- .enable = alsa_output_enable,
- .disable = alsa_output_disable,
- .open = alsa_open,
- .play = alsa_play,
- .drain = alsa_drain,
- .cancel = alsa_cancel,
- .close = alsa_close,
-
- .mixer_plugin = &alsa_mixer_plugin,
+ "alsa",
+ alsa_test_default_device,
+ alsa_init,
+ alsa_finish,
+ alsa_output_enable,
+ alsa_output_disable,
+ alsa_open,
+ alsa_close,
+ nullptr,
+ nullptr,
+ alsa_play,
+ alsa_drain,
+ alsa_cancel,
+ nullptr,
+
+ &alsa_mixer_plugin,
};
diff --git a/src/output/alsa_output_plugin.h b/src/output/AlsaOutputPlugin.hxx
index daa1f3615..dc7e639a8 100644
--- a/src/output/alsa_output_plugin.h
+++ b/src/output/AlsaOutputPlugin.hxx
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-#ifndef MPD_ALSA_OUTPUT_PLUGIN_H
-#define MPD_ALSA_OUTPUT_PLUGIN_H
+#ifndef MPD_ALSA_OUTPUT_PLUGIN_HXX
+#define MPD_ALSA_OUTPUT_PLUGIN_HXX
extern const struct audio_output_plugin alsa_output_plugin;
diff --git a/src/output/oss_output_plugin.c b/src/output/OssOutputPlugin.cxx
index e366a4537..5810825f1 100644
--- a/src/output/oss_output_plugin.c
+++ b/src/output/OssOutputPlugin.cxx
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -18,7 +18,7 @@
*/
#include "config.h"
-#include "oss_output_plugin.h"
+#include "OssOutputPlugin.hxx"
#include "output_api.h"
#include "mixer_list.h"
#include "fd_util.h"
@@ -60,7 +60,7 @@ struct oss_data {
struct audio_output base;
#ifdef AFMT_S24_PACKED
- struct pcm_export_state export;
+ struct pcm_export_state pcm_export;
#endif
int fd;
@@ -163,11 +163,10 @@ oss_output_test_default_device(void)
static struct audio_output *
oss_open_default(GError **error)
{
- int i;
int err[G_N_ELEMENTS(default_devices)];
enum oss_stat ret[G_N_ELEMENTS(default_devices)];
- for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
+ for (int i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
ret[i] = oss_stat_device(default_devices[i], &err[i]);
if (ret[i] == OSS_STAT_NO_ERROR) {
struct oss_data *od = oss_data_new();
@@ -182,7 +181,7 @@ oss_open_default(GError **error)
}
}
- for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
+ for (int i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
const char *dev = default_devices[i];
switch(ret[i]) {
case OSS_STAT_NO_ERROR:
@@ -243,7 +242,7 @@ oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
{
struct oss_data *od = (struct oss_data *)ao;
- pcm_export_init(&od->export);
+ pcm_export_init(&od->pcm_export);
return true;
}
@@ -252,7 +251,7 @@ oss_output_disable(struct audio_output *ao)
{
struct oss_data *od = (struct oss_data *)ao;
- pcm_export_deinit(&od->export);
+ pcm_export_deinit(&od->pcm_export);
}
#endif
@@ -504,7 +503,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
enum sample_format *sample_format_r,
int *oss_format_r,
#ifdef AFMT_S24_PACKED
- struct pcm_export_state *export,
+ struct pcm_export_state *pcm_export,
#endif
GError **error_r)
{
@@ -539,7 +538,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
*oss_format_r = oss_format;
#ifdef AFMT_S24_PACKED
- pcm_export_open(export, sample_format, 0, false, false,
+ pcm_export_open(pcm_export, sample_format, 0, false, false,
oss_format == AFMT_S24_PACKED,
oss_format == AFMT_S24_PACKED &&
G_BYTE_ORDER != G_LITTLE_ENDIAN);
@@ -556,16 +555,16 @@ static bool
oss_setup_sample_format(int fd, struct audio_format *audio_format,
int *oss_format_r,
#ifdef AFMT_S24_PACKED
- struct pcm_export_state *export,
+ struct pcm_export_state *pcm_export,
#endif
GError **error_r)
{
enum sample_format mpd_format;
enum oss_setup_result result =
- oss_probe_sample_format(fd, audio_format->format,
+ oss_probe_sample_format(fd, sample_format(audio_format->format),
&mpd_format, oss_format_r,
#ifdef AFMT_S24_PACKED
- export,
+ pcm_export,
#endif
error_r);
switch (result) {
@@ -603,7 +602,7 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
result = oss_probe_sample_format(fd, mpd_format,
&mpd_format, oss_format_r,
#ifdef AFMT_S24_PACKED
- export,
+ pcm_export,
#endif
error_r);
switch (result) {
@@ -635,7 +634,7 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format,
oss_setup_sample_rate(od->fd, audio_format, error_r) &&
oss_setup_sample_format(od->fd, audio_format, &od->oss_format,
#ifdef AFMT_S24_PACKED
- &od->export,
+ &od->pcm_export,
#endif
error_r);
}
@@ -749,14 +748,14 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
return 0;
#ifdef AFMT_S24_PACKED
- chunk = pcm_export(&od->export, chunk, size, &size);
+ chunk = pcm_export(&od->pcm_export, chunk, size, &size);
#endif
while (true) {
ret = write(od->fd, chunk, size);
if (ret > 0) {
#ifdef AFMT_S24_PACKED
- ret = pcm_export_source_size(&od->export, ret);
+ ret = pcm_export_source_size(&od->pcm_export, ret);
#endif
return ret;
}
@@ -771,18 +770,25 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
}
const struct audio_output_plugin oss_output_plugin = {
- .name = "oss",
- .test_default_device = oss_output_test_default_device,
- .init = oss_output_init,
- .finish = oss_output_finish,
+ "oss",
+ oss_output_test_default_device,
+ oss_output_init,
+ oss_output_finish,
#ifdef AFMT_S24_PACKED
- .enable = oss_output_enable,
- .disable = oss_output_disable,
+ oss_output_enable,
+ oss_output_disable,
+#else
+ nullptr,
+ nullptr,
#endif
- .open = oss_output_open,
- .close = oss_output_close,
- .play = oss_output_play,
- .cancel = oss_output_cancel,
-
- .mixer_plugin = &oss_mixer_plugin,
+ oss_output_open,
+ oss_output_close,
+ nullptr,
+ nullptr,
+ oss_output_play,
+ nullptr,
+ oss_output_cancel,
+ nullptr,
+
+ &oss_mixer_plugin,
};
diff --git a/src/output/oss_output_plugin.h b/src/output/OssOutputPlugin.hxx
index 2aecc2b3a..6c5c9530b 100644
--- a/src/output/oss_output_plugin.h
+++ b/src/output/OssOutputPlugin.hxx
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
* http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
-#ifndef MPD_OSS_OUTPUT_PLUGIN_H
-#define MPD_OSS_OUTPUT_PLUGIN_H
+#ifndef MPD_OSS_OUTPUT_PLUGIN_HXX
+#define MPD_OSS_OUTPUT_PLUGIN_HXX
extern const struct audio_output_plugin oss_output_plugin;
diff --git a/src/pcm_export.h b/src/pcm_export.h
index a7e7c3f68..005db48e0 100644
--- a/src/pcm_export.h
+++ b/src/pcm_export.h
@@ -87,6 +87,10 @@ struct pcm_export_state {
uint8_t reverse_endian;
};
+#ifdef __cplusplus
+extern "C" {
+#endif
+
/**
* Initialize a #pcm_export_state object.
*/
@@ -144,4 +148,8 @@ G_GNUC_PURE
size_t
pcm_export_source_size(const struct pcm_export_state *state, size_t dest_size);
+#ifdef __cplusplus
+}
+#endif
+
#endif
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-03 18:02:20 +0000